1. Field of the Invention
The present invention relates to a musical tone synthesizing apparatus which allows simple creation of sounds.
2. Background Art
Conventionally, there is known a technique wherein the sound generation mechanism of an acoustic instrument is simulated by a DSP (i.e., a Digital Sound Processor), etc., and whereby the musical tone of the instrument is composed. For example, a schematic algorithm of a physical model sound source, which has been utilized for simulation of acoustic wind instruments, will be described with reference to FIG. 7.
In FIG. 7, the numeral 101 designates a non-linear operation part which simulates the non-linear part of an acoustic wind instrument, namely, the reed. The numeral 102 designates a linear operation part which simulates the linear part of the acoustic wind instrument, namely, the resonance tube. The numeral 103 designates a digital-to-analog converter (hereinafter referred to as a "DAC") which measures pressure wave signals and other signals passing between the non-linear operation part 101 and the linear operation part 102. The DAC 103 further converts the measured signals to an analog signal and outputs the analog signal as a musical tone signal.
The non-linear operation part 101 receives a blow pressure signal PRE which represents the blowing pressure of the performer, and receives an embouchure signal EMB which represents the strain of lips of the performer, respectively, from external circuits. The non-linear operation part 101 further receives a reflected pressure wave signal q.sub.i from the linear operation part 102. The part 101 then generates a progressive pressure wave signal q.sub.0 in response to the received data and signals. The progressive pressure wave signal q.sub.0, generated by the non-linear operation part 101, is then supplied to the linear operation part 102 wherein the signal q.sub.0 is reflected and attenuated while passing through the part 102. As a result, the signal q.sub.0 is returned to the non-linear operation part 101 as a new reflected pressure wave signal q.sub.i.
Next, details of the linear operation part 102 will be described with reference to FIG. 8.
In the drawing, the numerals 113, 114 and 119 respectively, designate delay circuits which simulate the propagation delay of pressure waves occurred in the resonant tube of the acoustic wind instruments. The numerals 115 and 120, respectively designate low-pass filters (hereinafter referred to as "LPF") which respectively simulate the propagation losses of the pressure waves occurring in the resonance tube. The numerals 110 and 116 respectively designate multipliers, which respectively multiply the passing signals therein by reflection coefficients REFS and REFL, so as to simulate the losses occurring in the both ends of the resonance tube.
The numeral 200 designates a junction which simulates a tone hole provided in the resonant tube of acoustic wind instruments. The components described above are connected to each other so as to compose a loop circuit. The signals passing through the components simulate the progressive pressure waves and reflected pressure waves in the resonance tube.
The progressive pressure wave signal q.sub.0 generated by the non-linear operation part 101 is supplied to multiplier 121 and 118. In the multiplier 118, the supplied progressive pressure wave signal q.sub.0 is multiplied by an input gain constant NLSO, and the multiplication result is added to the reflected pressure wave signal at an adder 117. Therefore, influences applied to the reflected pressure wave by the blowing pressure can be simulated. Similarly, the progressive pressure wave signal q.sub.0 is multiplied by an input gain constant NLLO in a multiplier 122, the multiplication results thereof are added to the progressive pressure wave signal by an adder 112, and the influences applied to the progressive pressure wave by the blowing pressure can also be simulated.
Then, the progressive pressure wave signal generated via the LPF 120 and multiplier 110 is multiplied by an output gain NLSI of the linear part at a multiplier 111; the multiplication results thereof are then supplied to an adder 123. Similarly, the reflected pressure wave signal at the prior stage of adder 117 is multiplied by output gain NLLI of the linear part via a multiplier 121, and the multiplication results thereof are then supplied to the adder 123. The signals supplied to the adder 123 are then added by the adder 123, and the addition results are supplied to the non-linear operation part 101 as the reflected pressure wave signal q.sub.i.
As described above, in the linear operation part 102 shown in FIG. 8, the pressure wave signal is passed through the loop circuit consisting of delay circuits 113, 114, and 119, LPFs 115 and 120, etc., while being influenced by progressive pressure wave signal q.sub.o. As a result, the reflected pressure wave signal q.sub.i is formed and returned to the non-linear operation part 101.
Next, details of the non-linear operation part 101 will be described with reference to FIG. 9.
In FIG. 9, the numeral 140 designates a subtracter, subtracting the blow pressure signal PRE from the reflected pressure wave signal q.sub.i, generates the subtraction results as a difference pressure signal .DELTA.q. The difference pressure signal .DELTA.q is then supplied to a digital controlled filter 142 via a subtracter 141, simultaneously supplied to a Graham function table 148 via a multiplier 147, and simultaneously supplied to a repulsing function table 150.
The digital controlled filter 142, belonging to the secondary low-pass filter, has a cut-off frequency and an amplitude build-up ratio (Q) respectively given by variables LPFR and QLR. Furthermore, because the embouchure data EMB is supplied to the digital controlled filter 142, characteristics such as the cut-off frequency, etc., are set accordingly. The output signal of the digital controlled filter 142 is added with embouchure data EMB in adder 143. As a result, frequency response of the difference pressure signal .DELTA.q is set in response to conditions of performer's lips, so that the frequency setting mechanism of the performer's lips can be simulated.
Then, the output signal of adder 143 is weighted by means of being multiplied by a variable SLTGIN at a multiplier 144, and the weighted signal is then supplied to a slit function table 145. The slit function table 145 generates an opening area signal S.sub.L, representing the opening area of a performer's lips, in response to the difference pressure signal .DELTA.q which have been affected by the frequency response. This opening area signal S.sub.L is then supplied to the multiplier 149, and simultaneously multiplied by a feedback coefficient .beta. in the multiplier 146. The multiplication results are then returned to the subtracter 141.
As described above, according to the components 141.about.146 shown in FIG. 9, the opening area signal S.sub.L can be obtained in response to the difference pressure signal .DELTA.q, variables LPFR and QLR, and the embouchure data EMB.
Next, according to Graham's rule, the flux passing a unit area in a unit time, namely, air speed V can be expressed in the following formula (A1). EQU V=.sqroot.{2(.DELTA.q)/.rho.} (A1)
Herein, .rho. is the air density.
The Graham function table 148 gives the air speed signal V according to the formula (A1) when the difference pressure signal .DELTA.q is supplied. In the prior stage of Graham function table 148, a multiplier 147 is provided for adjusting the influences due to the Graham function, wherein the difference pressure signal .DELTA.q is multiplied with a prespecified variable GRMGIN (i.e., Graham gain).
As described above, when the opening area signal S.sub.L and the air speed signal V are obtained, they are multiplied in the multiplier 149, and the multiplication results is generated as a flow signal F.
The flow signal F is then supplied to a multiplier 153 wherein the signal F is multiplied by a variable Z which represents the input impedance of the resonance tube. Then, the multiplication results are generated as the progressive pressure wave signal q.sub.o via an adder 155.
Incidentally, in the acoustic lip-reed wind instruments, a so called "repulsing sound" may occur due to the pressure wave being reflected in the mouthpiece. In order to simulate the repulsing sound, the circuit shown in FIG. 9 is provided with a repulsing function table 150. When the difference pressure signal .DELTA.q is supplied to the table 150, a repulsing signal S.sub.RP, which represents the repulsing sound, is then generated. The generated repulsing signal S.sub.RP is then modified in a high-pass filter (hereinafter, referred to as HPF) 151 and supplied to a multiplier 152. The characteristics of the HPF 151 is set when a coefficient HPFR is supplied to the HPF 151.
The repulsing signal S.sub.RP, generated via the HPF 151, is then multiplied with the difference pressure signal .DELTA.q in the multiplier 152, is further weighted by the variable REPGIN in a multiplier 154, and is supplied to the adder 155. Therefore, influences due to the repulsing sound is are imparted to the progressive pressure wave signal q.sub.O.
FIGS. 10A and 10B respectively show the waveform of analog output signal generated by the DAC 103 and the frequency spectrum of the waveform analyzed by means of FFT (Fast Fourier Transform) analyzer. Furthermore, the waveform and frequency spectrum of opening area signal S.sub.L, air speed signal V, and repulsing signal S.sub.RP are respectively shown in FIGS. 11A to 13B.
Incidentally, in order to practice a sound creation on the above-described algorithm, various parameters (variables) must be varied appropriately. However, in some cases, it is difficult to estimate how the musical tone changes in response to the change of parameters. Accordingly, in these cases, it is difficult to clarify the correspondence between the tone color and other parameters. Furthermore, there are some parameters which make the sound signals go out of tune or make the modulation stop when the parameters are changed. Therefore, considerable skill is required by engineers to ultimately obtain the sounds by means of changing various parameters.
Furthermore, electronic musical instruments are, preferably, provided with various types of manually operable members in order to permit the performers to change various parameters during the performance. However, due to the above reasons, parameters chosen by the performer must be limited, and it is therefore difficult for him to improve expressiveness.